The present invention relates generally to apparatus and methods for adaptive feedback cancellation in an audio system such as a hearing aid and, more specifically, to a feedback cancellation system of the hearing aid with reduced sensitivity to low-frequency tonal inputs.
An audio system, such as a hearing aid, almost invariably incurs some sort of mechanical and/or acoustic feedback during operation of the audio system. The mechanical and/or acoustic feedback often limits the maximum gain that can be achieved in the hearing aid. Moreover, system instability caused by the feedback, whether mechanical and/or acoustic, is sometimes audible as a continuous high-frequency tone or whistle emanating from the hearing aid. The mechanical feedback of the hearing aid is usually caused by mechanical vibrations from a component thereof such as a receiver. Mechanical vibrations from the receiver of a high-power hearing aid can be reduced by combining outputs of two receiver units mounted back-to-back so as to cancel the net mechanical movement of the receiver units. As such, as much as 10 dB additional gain can be achieved for the high-power hearing aid before the onset of oscillation by the hearing aid. Many hearing aids also provide a venting capability to reduce unpleasant occlusion experienced by users of the hearing aids. But venting an earmold of a behind-the-ear (BTE) type hearing aid or a shell of an in-the-ear (ITE) type hearing aid establishes an acoustic feedback path that would limit the maximum possible gain to approximately less than 40 dB for a small vent and even less for a large vent. The acoustic feedback path includes effects from many of the hearing aid components such as the amplifier, receiver, and microphone as well as vent acoustics.
As mentioned, the acoustic feedback of the hearing aid tends to cause system instability of the hearing aid, particularly at high frequencies. A traditional approach for increasing the stability of a hearing aid is to reduce the gain at high frequencies. Reducing the gain of the hearing aid only at high frequencies modifies the overall system frequency response of the hearing aid. Therefore, controlling feedback by modifying the system frequency response to avoid instability means that a desired high-frequency response of the hearing aid will be sacrificed. Phase shifters and notch filters have also been suggested to control feedback, but have not proven to be very effective.
A more effective technique to control feedback is by feedback cancellation. For instance, an internal feedback signal is estimated and subtracted from a microphone signal of the hearing aid. Feedback cancellation typically uses an adaptive filter that models the dynamically changing feedback path of the hearing aid. Such an adaptive feedback cancellation system, however, can generate a large mismatch between an actual feedback path and an adaptive filter modeled feedback path when the input signal of the hearing aid is either narrowband or sinusoidal. One example of such a system has been disclosed by U.S. Pat. No. 5,091,952 to Williamson et al., as is illustrated in FIG. 1. FIG. 1 shows a hearing aid 100 having the adaptive feedback cancellation system incorporated therein. As shown in FIG. 1, an adaptive filter 101 is used to model the feedback path of the hearing aid, and a Least Mean Square (LMS) adaptation algorithm 103 is used to control filter coefficients adaptation of adaptive filter 101. A delay 105 is placed in the feedback path model to decorrelate the hearing aid output from the input. The delay 105 improves the system convergence of the hearing aid for signals such as speech. However, for tonal inputs at low frequencies such as music, sinusoids, or audiological test signals commonly used to measure hearing loss of a patient, this system tends to cancel the tonal inputs instead of accurately modeling the actual feedback path of the hearing aid for feedback cancellation.
An improved effective feedback cancellation scheme used in a hearing aid is disclosed by the present inventor in U.S. Pat. No. 6,072,884, entitled xe2x80x9cFeedback Cancellation Apparatus and Methodsxe2x80x9d, the contents of which are incorporated herein by reference. This improved system is illustrated in FIG. 2. The feedback path of such improved system is modeled by the combination of an adaptive filter 201 and a delay 205 plus a slowly-varying or non-varying (frozen) filter 219. The frozen filter 219 can be a frozen IIR filter or a frozen all pole filter, and the adaptive filter 201 can be an adaptive (all zero) FIR filter. Specifically, when the hearing aid is first turned on, filter (pole) coefficients of the frozen filter 219 are adapted to model those aspects of the feedback path that can have high-Q resonance but which stay relatively constant during normal hearing aid operation. Thus, pole coefficients of the feedback path, once determined, are modified and then frozen or, at least, changed vary slowly. Once the pole coefficients are determined, filter (zero) coefficients of the adaptive filter 201 are adapted to correspond to the modified poles. The objective of this adaptation is to minimize an error signal e(n) produced at the output of adder 209. Unlike the filter coefficients of the frozen filter 219, the adaptive filter 201 continues to adapt its filter coefficients in response to changes in the feedback path. Therefore, the adaptive filter 201 models those portions of the feedback path that are changing, and the frozen filter 219 models those portions of the feedback path that remain essentially constant while the hearing aid is in use. This improved system will, however, also attempt to cancel a tonal input signal. Nonetheless, adaptive filter coefficients of this improved system are constrained to prevent excessive deviation from an initial setting thereof. In the presence of a tonal input, the degree of input signal cancellation resulting from the adaptive filter is greatly reduced, but it is still not completely eliminated.
The feedback cancellation systems shown in FIGS. 1 and 2 use the LMS algorithm for adaptation of the adaptive filter coefficients. As shown in FIGS. 1 and 2, the hearing aid receives an input signal x(n), a transfer function of a hearing aid processing unit is given by h(n), and the hearing aid output is y(n), where n is a sample index. The LMS algorithm adaptation in both the above-mentioned feedback cancellation systems uses the cross-correlation of an error signal e(n) and a feedback path signal d(n) that is inputted to the adaptive filter (i.e., the adaptive filter 101 or the adaptive filer 201). The objective of the adaptive filter is to minimize the power of the error signal e(n). Let the adaptive filter be a K-tap finite impulse response (FIR) filter having adaptive coefficients bl(n) through bk(n), a power-normalized adaptive filter update for input sample index n is then given by                                           b            k                    ⁡                      (                          n              +              1                        )                          =                                            b              k                        ⁡                          (              n              )                                +                      2            ⁢                          μ                                                σ                  d                  2                                ⁡                                  (                  n                  )                                                      ⁢                          e              ⁡                              (                n                )                                      ⁢                          d              ⁡                              (                                  n                  -                  k                                )                                                                        (        1        )            
where xcexc controls the rate of adaptation and "sgr"d2(n) is the average power in the feedback path signal d(n). If the input signal x(n) is white noise, the adaptive filter will normally converge to a model of its feedback path. If the input x(n) is a pure tone, however, the adaptive feedback cancellation system will minimize the error signal e(n) by adjusting the filter coefficients bl(n) through bk(n) so that v(n), which is an adaptively filtered version of d(n), has the same amplitude and phase as of the input x(n) and thus will cancel the tone. Slowing the rate of adaptation by making xcexc smaller will reduce the tendency to cancel short-duration tonal inputs, but will also reduce the ability of the adaptive system to rapidly adapt to large changes to the acoustic feedback path.
A further improvement in feedback cancellation for hearing aids is disclosed by Gao et al. in an international patent application WO 00/019605 A2. This system is illustrated in FIG. 3. As shown in FIG. 3, its feedback path is modeled by the combination of an adaptive filter 301, a delay 305, an LMS adaptation 303, and a frozen filter 319, as previously taught by the above-mentioned ""884 patent. In FIG. 3, however, both inputs to the LMS adaptation 303 used to update the adaptive filter coefficients are further filtered through fixed filters p(n) 321 and 323. The fixed filters p(n) 321, 323 are bandpass or highpass filters, and emphasize a frequency region where mismatch between the actual and modeled feedback paths can cause the greatest stability problems in the hearing aid. Low frequencies, where the hearing aid typically has low gain but where tonal input signals are often experienced, are de-emphasized to minimize the possibility of canceling a tonal input. This further improved system relies on the fixed filters p(n) 321, 323 to reduce the potential mismatch when a tonal input is present, and the filter adaptation is not constrained.
In the system of FIG. 3, the cancellation of tonal input signals is reduced by minimizing the power in a filtered version of the error signal instead of minimizing the broadband error. The inputs g(n) and f(n) to LMS adaptation 303 are passed through the respective fixed filters p(n) 321, 323 giving g(n)=e(n)*p(n) and f(n)=d(n)*p(n), where * denotes convolution by the fixed filters p(n) 321, 323. The adaptive coefficient update for input sample n is then given by:                                                         b              k                        ⁡                          (                              n                +                1                            )                                =                                                    b                k                            ⁡                              (                n                )                                      +                          2              ⁢                              μ                                                      σ                    f                    2                                    ⁡                                      (                    n                    )                                                              ⁢                              g                ⁡                                  (                  n                  )                                            ⁢                              f                ⁡                                  (                                      n                    -                    k                                    )                                                                    ,                            (        2        )            
where xcexc controls the rate of adaptation and "sgr"f2(n) is the average power in signal f(n). The use of a highpass filter for p(n), for example, is equivalent to making xcexc smaller at low frequencies, thus slowing the rate of adaptation for low-frequency input signals. However, even the system shown in FIG. 3 will tend to cancel a tonal input at low frequencies if the signal duration is long enough.
A need, thus, remains in the art for apparatus and methods to reduce the cancellation of tonal input signals when implementing adaptive feedback cancellation in a hearing aid or other audio system.
A feedback cancellation system with reduced sensitivity to low-frequency tonal inputs is provided. Such a system can be used, for example, in a hearing aid to prevent cancellation of the desired tonal inputs to the hearing aid, thus improving the gain at high frequencies while simultaneously preserving the desired tonal inputs at low frequencies. The feedback cancellation system comprises a first adaptive filter block for adaptively filtering an error signal to remove the low-frequency tonal components from the error signal. The first adaptive filter block is constrained so that only low-frequency tones in the error signal are cancelled, thus enabling the feedback cancellation system to still cancel xe2x80x9cwhistlingxe2x80x9d at high frequencies due to the temporary instability of the hearing aid. A second adaptive filter block adaptively filters the feedback path signal to produce an adaptively filtered feedback path signal. The first and second adaptive filter blocks are identical and filter coefficients of the first adaptive filter block are copied to those of the second adaptive filter block. Using an LMS adaptation algorithm, filter coefficients of the adaptive filer of the feedback cancellation system are controlled by the adaptively filtered error signal and the adaptively filtered feedback path signal respectively inputted from the first and second adaptive filter blocks. The adaptive filter then produces an adaptively filtered modeled feedback signal to be subtracted from an electrical audio signal input for updating the error signal of the hearing aid. The hearing aid processes the updated error signal with a digital signal processor to generate an audio output.
Thus, in one aspect, the invention is an audio processing system such as used in a hearing aid, the audio processing system comprised of a signal path including a digital signal processing means for processing an error signal, and a feedback cancellation means that adaptively models an acoustic feedback path. The feedback cancellation means includes first adaptive filter means adaptively filtering the error signal to remove low-frequency tonal components of the error signal for coefficient adaptation of the acoustic feedback path model, an LMS adaptation means, and an adaptive filter. The filter coefficients of the adaptive filter are adaptively controlled by the adaptively filtered error signal to produced an adaptive feedback signal. Preferably, the signal path of the audio processing system is also comprised of an input transducer, a subtracting means, and an output transducer. In a preferred embodiment, the first adaptive filter means comprises at least one adaptive notch filter. If more than one adaptive notch filters are included in the first adaptive filter means, they are connected in cascade to each other. In another preferred embodiment, the first adaptive filter means comprises a fixed bandpass filter filtering the error signal and connected in cascade to the at least one adaptive notch filter. In yet another preferred embodiment, the first adaptive filter means comprises a fixed highpass filter filtering the error signal and connected in cascade to the at least one adaptive notch filter. In yet another preferred embodiment, the first adaptive filter means comprises a plurality of bandpass filters arranged in parallel combination and respectively receiving the error signal, a plurality of adaptive notch filters also arranged in parallel combination, and adder means for summing outputs of the plurality of adaptive notch filters. Each of the plurality of adaptive notch filters is connected to the output of one of the plurality of bandpass filters. In yet another preferred embodiment, the first adaptive filter means comprises a highpass filter filtering the error signal, a lowpass filter filtering the error signal, a delay delaying the output of the lowpass filter, an adaptive FIR filter adaptively filtering the output of the delay, a first subtracting means for subtracting the output of the adaptive FIR filter from the output of the lowpass filter, and a first adder means for summing the output of the first subtracting means and the output of the highpass filter.
In another aspect, the invention is an audio processing system such as used in a hearing aid, the audio processing system comprised of a signal path including a digital signal processing means for processing an error signal, and a feedback cancellation means that adaptively models an acoustic feedback path. The feedback cancellation means includes first adaptive filter means adaptively filtering the error signal to remove low-frequency tonal components of the error signal for coefficient adaptation of the acoustic feedback path model, second adaptive filter means for adaptive filtering a feedback path signal, an LMS adaptation means, and an adaptive filter. The filter coefficients of the adaptive filter are adaptively controlled by the adaptively filtered error signal and by the adaptively filtered feedback path signal to produced an adaptive feedback signal. The first and second adaptive filter means are identical and filter coefficients of first adaptive filter means are copied to those of the second adaptive filter means. Preferably, the signal path of the audio processing system is also comprised of an input transducer, a subtracting means, and an output transducer. In a preferred embodiment, the first adaptive filter means comprises at least one adaptive notch filter. If more than one adaptive notch filters are included in the adaptive filter means, they are connected in cascade to each other. In another preferred embodiment, the first adaptive filter means comprises a fixed bandpass filter filtering the error signal and connected in cascade to the at least one adaptive notch filter. In yet another preferred embodiment, the first adaptive filter means comprises a fixed highpass filter filtering the error signal and connected in cascade to the at least one adaptive notch filter. In yet another preferred embodiment, the first adaptive filter means comprises a plurality of bandpass filters arranged in parallel combination and respectively receiving the error signal, a plurality of adaptive notch filters also arranged in parallel combination, and adder means for summing outputs of the plurality of adaptive notch filters. Each of the plurality of adaptive notch filters is connected to the output of one of the plurality of bandpass filters. In yet another preferred embodiment, the first adaptive filter means comprises a highpass filter filtering the error signal, a lowpass filter filtering the error signal, a delay delaying the output of the lowpass filter, an adaptive FIR filter adaptively filtering the output of the delay, a first subtracting means for subtracting the output of the adaptive FIR filter from the output of the lowpass filter, and a first adder means for summing the output of the first subtracting means and the output of the highpass filter.
In yet another aspect, the invention is a method of feedback cancellation, such as used in a hearing aid, the method comprising the steps of receiving an input signal, generating an electrical audio signal in accordance with the input signal, processing the electrical audio signal by a digital signal processor to produce an electrical output signal, estimating an internal feedback signal in accordance with the electrical output signal, generating an error signal by subtracting the internal feedback signal from the electrical audio signal, adaptively filtering the error signal to remove low-frequency tonal components of the error signal with a first adaptive filter block, adaptively controlling filter coefficients of an adaptive filter in accordance with the adaptively filtered error signal, updating the internal feedback signal by the adaptive filter, updating the error signal by subtracting the updated internal feedback signal from the electrical audio signal, and processing the updated error signal by the digital signal processor to update the electrical output signal. In a preferred embodiment, the step of adaptively filtering the error signal is accomplished by filtering the error signal with at least one adaptive notch filter of the first adaptive filter block. In another embodiment, the step of adaptively filtering the error signal is accomplished by filtering the error signal with a bandpass filter and then with the at least one adaptive notch filter. In yet another embodiment, the step of adaptively filtering the error signal is accomplished by filtering the error signal with a highpass filter and then with the at least one adaptive notch filter. In yet another embodiment, the step of adaptively filtering the error signal comprises the steps of filtering the error signal with a plurality of bandpass filters arranged in parallel combination, filtering outputs of the plurality of bandpass filters with a plurality of adaptive notch filters also arrange in parallel combination, and generating the adaptively filtered error signal by summing outputs of the plurality of adaptive notch filters. In yet another embodiment, the step of adaptively filtering the error signal comprises the steps of generating a highpass error signal by filtering the error signal with a highpass filter, generating a lowpass filtered error signal by filtering the error signal with a lowpass filter, delaying the lowpass filtered error signal, generating an adaptively filtered lowpass error signal by filtering the delayed lowpass filtered error signal with an adaptive FIR filter, generating a lowpass error signal by subtracting the adaptively filtered lowpass error signal from the lowpass filtered error signal, and generating the adaptively filtered error signal by summing the lowpass error signal and the highpass error signal.
In yet another aspect, the invention is a method of feedback cancellation, such as used in a hearing aid, the method comprising the steps of receiving an input signal, generating an electrical audio signal in accordance with the input signal, processing the electrical audio signal by a digital signal processor to produce an electrical output signal, estimating an internal feedback signal in accordance with the electrical output signal, generating an error signal by subtracting the internal feedback signal from the electrical audio signal, adaptively filtering the error signal to remove low-frequency tonal components of the error signal with a first adaptive filter block, delaying the electrical output signal with a delay unit, generating a feedback path signal by filtering an output of the delay unit with a frozen filter, generating an adaptive feedback path signal by filtering the feedback path signal with a second adaptive filter block, adaptively controlling filter coefficients of an adaptive filter in accordance with the adaptively filtered error signal and the adaptively filtered feedback path signal, updating the internal feedback signal by the adaptive filter, updating the error signal by subtracting the updated internal feedback signal from the electrical audio signal, and processing the updated error signal by the digital signal processor to update the electrical output signal. In a preferred embodiment, the step of adaptively filtering the error signal is accomplished by filtering the error signal with at least one adaptive notch filter of the first adaptive filter block. In another embodiment, the step of adaptively filtering the error signal is accomplished by filtering the error signal with a bandpass filter and then with the at least one adaptive notch filter. In yet another embodiment, the step of adaptively filtering the error signal is accomplished by filtering the error signal with a highpass filter and then with the at least one adaptive notch filter. In yet another embodiment, the step of adaptively filtering the error signal comprises the steps of filtering the error signal with a plurality of bandpass filters arranged in parallel combination, filtering outputs of the plurality of bandpass filters with a plurality of adaptive notch filters also arrange in parallel combination, and generating the adaptively filtered error signal by summing outputs of the plurality of adaptive notch filters. In yet another embodiment, the step of adaptively filtering the error signal comprises the steps of generating a highpass error signal by filtering the error signal with a highpass filter, generating a lowpass filtered error signal by filtering the error signal with a lowpass filter, delaying the lowpass filtered error signal, generating an adaptively filtered lowpass error signal by filtering the delayed lowpass filtered error signal with an adaptive FIR filter, generating a lowpass error signal by subtracting the adaptively filtered lowpass error signal from the lowpass filtered error signal, and generating the adaptively filtered error signal by summing the lowpass error signal and the highpass error signal.
A further understanding of the nature and advantages of the present invention may be realized by reference to the remaining portions of the specification and the drawings.